From dc58831c27336536626a67c8148e645265045f40 Mon Sep 17 00:00:00 2001
From: Danilo Bargen <mail@dbrgn.ch>
Date: Thu, 12 May 2016 15:47:19 +0200
Subject: [PATCH] Update RTCConfiguration interface to latest spec

---
 webrtc/RTCPeerConnection-tests.ts | 24 +++++++++++++++++--
 webrtc/RTCPeerConnection.d.ts     | 39 +++++++++++++++++++++++++++----
 2 files changed, 56 insertions(+), 7 deletions(-)

diff --git a/webrtc/RTCPeerConnection-tests.ts b/webrtc/RTCPeerConnection-tests.ts
index ea850f8724..9cd8087947 100644
--- a/webrtc/RTCPeerConnection-tests.ts
+++ b/webrtc/RTCPeerConnection-tests.ts
@@ -3,8 +3,28 @@
 
 let voidpromise: Promise<void>;
 
-var config: RTCConfiguration =
-    { iceServers: [{ urls: "stun.l.google.com:19302" }] };
+var minimalConfig: RTCConfiguration = {};
+var config: RTCConfiguration = {
+    iceServers: [
+        {
+            // Single url
+            urls: "stun.l.google.com:19302"
+        },
+        {
+            // List of urls and credentials
+            urls: ["another-stun.example.com"],
+            username: "dude",
+            credential: "pass",
+            credentialType: "token"
+        },
+    ],
+    iceTransportPolicy: "relay",
+    bundlePolicy: "max-compat",
+    rtcpMuxPolicy: "negotiate",
+    peerIdentity: "dude",
+    certificates: [{expires: 1337}],
+    iceCandidatePoolSize: 5
+};
 var constraints: RTCMediaConstraints =
     { mandatory: { offerToReceiveAudio: true, offerToReceiveVideo: true } };
 
diff --git a/webrtc/RTCPeerConnection.d.ts b/webrtc/RTCPeerConnection.d.ts
index dcd36d9aec..a3d39130e0 100644
--- a/webrtc/RTCPeerConnection.d.ts
+++ b/webrtc/RTCPeerConnection.d.ts
@@ -20,17 +20,46 @@
 // TODO(2): get Typescript to have union types as WebRtc uses them.
 // https://typescript.codeplex.com/workitem/1364
 
-interface RTCConfiguration {
-  iceServers: RTCIceServer[];
+// https://www.w3.org/TR/webrtc/#idl-def-RTCIceTransportPolicy
+type RTCIceTransportPolicy = 'public' | 'relay' | 'all';
+
+// https://www.w3.org/TR/webrtc/#idl-def-RTCBundlePolicy
+type RTCBundlePolicy = 'balanced' | 'max-compat' | 'max-bundle';
+
+// https://www.w3.org/TR/webrtc/#idl-def-RTCRtcpMuxPolicy
+type RTCRtcpMuxPolicy = 'negotiate' | 'require';
+
+// https://www.w3.org/TR/webrtc/#idl-def-RTCCertificate
+interface RTCCertificate {
+    expires: number;
 }
+
+// https://www.w3.org/TR/webrtc/#idl-def-RTCConfiguration
+// https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/RTCPeerConnection#RTCConfiguration_dictionary
+interface RTCConfiguration {
+  iceServers           ?: RTCIceServer[];        // optional according to mozilla docs
+  iceTransportPolicy   ?: RTCIceTransportPolicy; // default = 'all'
+  bundlePolicy         ?: RTCBundlePolicy;       // default = 'balanced'
+  rtcpMuxPolicy        ?: RTCRtcpMuxPolicy;      // default = 'require'
+  peerIdentity         ?: string;                // default = null
+  certificates         ?: RTCCertificate[];      // default is auto-generated
+  iceCandidatePoolSize ?: number;                // default = 0
+}
+
 declare var RTCConfiguration: {
   prototype: RTCConfiguration;
   new (): RTCConfiguration;
 };
 
+// https://www.w3.org/TR/webrtc/#idl-def-RTCIceCredentialType
+type RTCIceCredentialType = 'password' | 'token';
+
+// https://www.w3.org/TR/webrtc/#idl-def-RTCIceServer
 interface RTCIceServer {
-  urls: string;
+  urls: string | string[];
+  username?: string;
   credential?: string;
+  credentialType?: RTCIceCredentialType; // default = 'password'
 }
 declare var RTCIceServer: {
   prototype: RTCIceServer;
@@ -42,7 +71,7 @@ interface mozRTCPeerConnection extends RTCPeerConnection {
 }
 declare var mozRTCPeerConnection: {
   prototype: mozRTCPeerConnection;
-  new (settings: RTCPeerConnectionConfig,
+  new (settings?: RTCConfiguration,
        constraints?:RTCMediaConstraints): mozRTCPeerConnection;
 };
 // webkit (Chrome) specific prefixes.
@@ -50,7 +79,7 @@ interface webkitRTCPeerConnection extends RTCPeerConnection {
 }
 declare var webkitRTCPeerConnection: {
   prototype: webkitRTCPeerConnection;
-  new (settings: RTCPeerConnectionConfig,
+  new (settings?: RTCConfiguration,
        constraints?:RTCMediaConstraints): webkitRTCPeerConnection;
 };